Asterisk Forums

I am not an expert in asterisk, but i had a similar problem. in my case what happened was i installed asterisk before installing dahdi-linux and dahdi-tool. so my asterisk compiled without dahdi drivers. so just make sure u install dahdi and the lib drivers first before u install the asterisk.
I hope this helps
regards

Im new to AsteriskNOW. Ive studied the Avaya Hardware PBX's and Im able to get them to work no problem. However Im unable to transfer those skills to the Asterisk system.

First things to note is what im using
Im using VMWare to host a Virtual PBX using AsteriskNOW 2.0
The Server's Ip Address is 192.168.1.105 (LAN ip of course as im playing with this server in a Lan only environment)
XLite Softphone v3.0

What I can Do
I can Connect to the Web interface and set up the extentions
I can Ping 192.168.1.105

What I cant do
Set any sort of dialplan analysis table
Have the Xlite Softphone on the same network register itself
Diagnose or debug any soft of communications. I really have no idea how to narrow down where the communications breakdown is.

Having installed AsteriskNOW i Realized that it doesnt actually use asterisk, and it uses FreePBX. Many of the guides availiable is for the Asterisk pbx only. while the webadmin is working from other networked machines, I cannot actually access the pbx server on the PBX VMware Image. (Mind you i have no idea where it even is)

Im an amature when it comes to linux. Ive had some schooling in it, but not extensive. I dont know how to access the configuration of AsteriskNOW in the CentOS to type commands as ive seen people suggest to help with diagnostics in some other posts.

In command prompt ive been successful in pinging 192.168.1.105 and since i can access the web interface, My assumption is that the ports required for use of the pbx is open as well. I would assume the firewall is disabled as i didnt enable it when I had first installed it. The AsteriskNOW VM image seems to be CLI only with no GUI and again being an amature Linux User, Im unable to check the status of the firewall, etc.

Could someone please help me acheive connectivity between the PBX and my networked softphone?

I installed B410P. All is OK....
When I want configure chan_dahdi.conf and defined "channel => x,x", asterisk show
"No such command 'dahdi show channels' "

Next I commented like ";channel => x,x" in chan_dahdi.conf, asterisk show

"Chan Extension Context Language MOH Interpret
pseudo default default"

chan_dahdi.conf

--------------------------------------------------------------------------
[trunkgroups]

[channels]
;language = cz

; include dahdi extensions defined in FreePBX
;#include chan_dahdi_additional.conf
;#include dahdi-channels.conf

group=0,12
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 4-5
;dahdichan = 4
context = default
group = 63

system.conf

------------------------------------------------------------------------------

1. Autogenerated by /usr/sbin/dahdi_genconf on Sun Oct 11 19:31:23 2009
2. If you edit this file and execute /usr/sbin/dahdi_genconf again,
3. your manual changes will be LOST.
4. Dahdi Configuration File

#
# This file is parsed by the Dahdi Configurator, dahdi_cfg

#
# Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) AMI/CCS RED
span=1,1,0,ccs,ami

1. termtype: te
bchan=1-2
hardhdlc=3
echocanceller=mg2,1-2
2. Span 2: B4/0/2 "B4XXP (PCI) Card 0 Span 2" AMI/CCS
span=2,2,0,ccs,ami
3. termtype: te
bchan=4-5
hardhdlc=6
echocanceller=mg2,4-5
4. Span 3: B4/0/3 "B4XXP (PCI) Card 0 Span 3" AMI/CCS RED
span=3,3,0,ccs,ami
5. termtype: te
bchan=7-8
hardhdlc=9
echocanceller=mg2,7-8
6. Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" AMI/CCS
span=4,4,0,ccs,ami
7. termtype: te
bchan=10-11
hardhdlc=12
echocanceller=mg2,10-11
8. Global data

loadzone = us
defaultzone = us

--------------------------------------------------------------------

Version of Asterisk 1.4.24
Driver DAHDI 2.2.0.2
kernel 2.6.18-164.el5
OS centos 5.3

Can you help me.
Thanks

You say you have port 5060 and friends "open" on the router... but in both directions? You must specifically port-forward the incoming traffic on those ports to 192.168.254.6 (your Asterisk server).

hello

I am running asterisk 1.4 and i am new. I have 2 broadvoice accounts. I want to configure my dial plan in such a way that when a user wants to make a call and line1 is busy, he should be connected to line2 to make the call and if both lines are busy there should be a playback telling him to wait.

for now what is happening is that they can only use line1 making the line2 useless

thanks


this is my current extensions.conf


[general]
static=yes
writeprotect=yes
autofallthrough=yes
clearglobalvars=no
;priorityjumping=yes

[globals]



[default]

include => sip1
include => sip2




[sip1]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@9036199554,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()

[sip2]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@2402185894,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()

i am a beggining programmer who catches on well to guidence by others i am looking for a fellow programmer to guide me with my scripting in orderfor me to learn. i use already the MIRC community although its not as helpfull as i need. if anyone is willing to help a begginer asterisk programmer please contact me , thanks!
YUda

Hi,

upgraded asterisk from 1.4.19 to 1.6.1, and now getting something like this:

Quote:
[Oct 11 13:21:46] ERROR[4039]: chan_dahdi.c:8793 mkintf: Signalling requested on channel 109 is ISDN PRI but line is in Unknown signalling 524416 signalling
[Oct 11 13:21:46] ERROR[4039]: chan_dahdi.c:14170 build_channels: Unable to register channel '94-124'


Any idea how to fix it?

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Regards
tahir almas
http://www.ictinnovations.co

We are pleased to annourse that We have added Fax broadcasting capability in our Solution for ITSP named "ICT Call Generator"

ICT Call Generator is voice and fax broadcasting platform based on asterisk switching engine , it support multiuser , multi compaign , multi technologies sip , iax , dahdi and multi codec with advacne autodialer features

for detail features and demo please visit following link

http://www.ictinnovations.com/voice-broadcasting

Regards

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Ok folks, having a bit of trouble getting softphones to work on my asterisk configuration.

Here's how everything is set up:

192.168.254.254 - Modem/Router (internal IP address)
192.168.254.5 - Web Server
192.168.254.6 - Asterisk Server
192.168.254.7 - Linksys PAPT2 (extensions 501 and 502)

/etc/asterisk/sip.conf:
Code: [general]
context=general
useragent=Asterisk PBX
bindport=5060
nindaddr=0.0.0.0
language=en
disallow=all
disable=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
nat=yes

[501]
type=friend
context=phones
host=dynamic

[502]
type=friend
context=phones
host=dynamic


On the modem/router I have TCP ports 5060, and 10000 through 20000 open, along with UDP port 5060. What I'd like to do is set up extensions 503 - 509 on softphones (like X-Lite) from external locations (i.e. non-192.168.254.x IP addresses). However, every time I try with these settings, I get an error 408, "Registration Timeout" on the softphone side, and on the asterisk side (verbosity set to 10), I get nothing. The attempted registration doesn't even show up.

Are the wrong ports open, or do I need to change my configuration somehow?

Thanks!

Rebuild the gui from source and it should give you the full URL.

Ok, other Linux/Ekiga/Twinkle users can conference in just fine. But when someone is using xlite on windows, they do not even get an answer, and the terminal only gives this output:

Code:  
hate*CLI>
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6


I looked at an example sip.conf entry for xlite, and mocked it, and it made no difference (the settings were almost the same anyway). I'll blame windows on this one

The funny thing is, it just started working, without making any modifications... I will update the next time this happens with code.

Ok, it works great for me, calling in from local sip to local asterisk. Others can call me, but they could not call into the conference room like i can, using the same number? Any clues?

Ok, it works great for me, calling in from local sip to local asterisk. Others can call me, but they could not call into the conference room like i can, using the same number? Any clues?

pastebin the output from asterisk -vvvvvvvvvvr while others are trying to call in.

well, for my local machine, i always keep it updated. But when i pay to upgrade my server space for my websites, which will allow me to host other servers as well, i will not touch what is not broken

You can always set yum to not update the kernel* then you dont have to worry about it. I generally dont do to many updates on my system once I get it up and running, unless its a security update. I figure why mess with whats working.